1. Field of the Invention
The invention concerns an intelligent acoustic microphone front end and a process for operation thereof according to the precharacterizing portion of Patent claims 1 and 6.
Voice command operated systems are being installed in modern motor vehicles with increasing frequency. Such systems should be operable by various vehicle occupants and from various seating positions. It is also an object, in the framework of modern occupant communication systems, to equip the individual seat places in the passenger compartment with specifically assigned microphones and loudspeakers. Thereby it is to be ensured that the individual occupants can communicate comfortably at any time from any position, independent of the physical design of the occupant communication system.
2. Related Art of the Invention
Japanese document JP 2002-091469 discloses a speech recognition system using a switchable, directionally selective microphone array for beam-forming for recording speech signals. The unit for beam-forming includes a directional recognizer, which uses the recorded speech signals to recognize the azimuth of the direction of speech origin. For this, signals are received from different angular directions with regard to the reception characteristic of the microphone array and subjected to a sound level analysis. In the framework of this threshold analysis the direction recognizer provides the most probable direction from which the speech signal originates as recognition result. This result is then used by the unit for beam forming for controlling the microphone array, so that the reception diagram of the microphone array is oriented with the greatest probability as close as possible towards the direction from which the speech signal originates.
From published EP 1 081 682 A2 a speech recognition system for an occupant communication system is known, which includes a number of differently positioned microphone systems. The signals of the individual microphone systems are detected in parallel and analyzed with regard to the signal-noise ratio parameter and the average speech level contained in the signal. Those signals, of which the parameters exceed predetermined threshold values, are supplied to a speech recognizer, which sequentially processes the individual signals, beginning with the strongest signal. From the positive recognition results, the speech content of the signal is determined. By the parallel analysis of the microphone signals recorded at different locations, a speaker is no longer required to speak the voice commands in a particular direction; it is also possible for the direction of speech to change during the speaking process, without loss of information.
A further speech recognition setup for parallel processing of multiple parallel microphones channels is described in German laid open DE 100 30 105 A1. Here the individual microphone channels are subjected to a character extraction independent from each other, and merged for the first time in the framework of a common classification. On this basis the information supplied to the classification still contains the information associated with the individual channels. It is also possible to individually carry out the preliminary processing in the individual channels, and to independently monitor the therefrom resulting effects on the signals.